Re: [asterisk-users] Call Queue advise
ok thanks. The problem isn't agents not wanting to get calls. Then they just wont answer. Some slower to answer users are complaining the people are not working to be able to answer the call quicker. @Paul Belanger - Option two is where I am thinking but I am trying to figure out the best way to do so. my current thought is a Macro the queue runs at the answer of a call. Currently agents do not log in and out. That is slatted for roll out after a few more hires to that team. Thanks, -Bryan Anderson On Wed, Dec 11, 2013 at 10:22 AM, Chad Wallace cwall...@lodgingcompany.comwrote: On Mon, 9 Dec 2013 16:15:14 -0800 Bryan Anderson shadow...@gmail.com wrote: On Mon, Dec 9, 2013 at 4:11 PM, Chad Wallace cwall...@lodgingcompany.comwrote: On Mon, 9 Dec 2013 15:47:57 -0800 Bryan Anderson shadow...@gmail.com wrote: I have a call queue that rings about 15 users and they are wanting to set it up so that the last person to answer a call doesn't ring on the next incoming call. Wouldn't the leastrecent strategy work for that? It wouldn't absolutely forbid an agent from taking the next call, but it would make sure every other agent had priority. You could also add a large wrap up time, to ensure they never get a second call within a certain time period. yes but I believe that least recent would ring one agent at a time? If my understanding is incorrect please correct it. We are wanting to keep with multiple phones ring to ensure coverage. Yes, you're right. It seems your solution (setting a penalty after they get a call) is probably the only one--unless, as has been suggested, you rethink your requirements. You could also remove them from the queue or pause them instead of changing their penalty. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Queue advise
I have a call queue that rings about 15 users and they are wanting to set it up so that the last person to answer a call doesn't ring on the next incoming call. What would be the best way to handle this? I have been looking at the strategies and none of those seem to be right for this. My current thoughts are probably a macro that places a penalty on the user tell the next call is answered. Any advice for this would be greatly appreciated. Thanks, -Bryan Anderson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Queue advise
yes but I believe that least recent would ring one agent at a time? If my understanding is incorrect please correct it. We are wanting to keep with multiple phones ring to ensure coverage. Thanks, Bryan -Bryan Anderson On Mon, Dec 9, 2013 at 4:11 PM, Chad Wallace cwall...@lodgingcompany.comwrote: On Mon, 9 Dec 2013 15:47:57 -0800 Bryan Anderson shadow...@gmail.com wrote: I have a call queue that rings about 15 users and they are wanting to set it up so that the last person to answer a call doesn't ring on the next incoming call. What would be the best way to handle this? I have been looking at the strategies and none of those seem to be right for this. My current thoughts are probably a macro that places a penalty on the user tell the next call is answered. Any advice for this would be greatly appreciated. Wouldn't the leastrecent strategy work for that? It wouldn't absolutely forbid an agent from taking the next call, but it would make sure every other agent had priority. You could also add a large wrap up time, to ensure they never get a second call within a certain time period. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Homer SipCapture
Is anyone using Homer from sipcapture.org or anything like it for capture sip traffic for debuging? If so what are your experiences. Thanks -Bryan Anderson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Users.conf vs Sip.conf
I am using asterisk 1.8.5.0 and I am curious as to the roles of sip.conf and users.conf. My understanding is to provision phones you use users.conf. Doing so creates a user, and a phone profile. With that said my understanding is that sip.conf is the prefered method for creating sip accounts since it provides more flexibility. If I could get some help/clarification/advice as to the ideal setup between the two files to: 1) create users and, 2) provision phones I would greatly appreciate it. Thanks, -Bryan Anderson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SPIP config
Ok, thanks for the info. -Bryan Anderson On Thu, Mar 7, 2013 at 6:07 PM, Chad Wallace cwall...@lodgingcompany.comwrote: On Thu, 7 Mar 2013 17:12:47 -0800 Bryan Anderson shadow...@gmail.com wrote: Has any one ever worked with placing idle display images onto the Polycom SPIP331 phones? I have got it working but when the image is displayed the clock is moved to the top of the screen. That is great but it scrolls between the clock and the registered extension(s) . Has anyone figured out a way to stop the scrolling and just display the time? If so could you provide me the configuration parameter? Sorry to say... we have the same problem with the 321s. Never managed to figure it out. I asked Polycom about it, and they said we'd have to get our vendor to order it as a feature request, or something like that. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom SPIP config
Has any one ever worked with placing idle display images onto the Polycom SPIP331 phones? I have got it working but when the image is displayed the clock is moved to the top of the screen. That is great but it scrolls between the clock and the registered extension(s) . Has anyone figured out a way to stop the scrolling and just display the time? If so could you provide me the configuration parameter? thanks, Bryan Anderson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DST offset
Hello,
I am trying to set the tcpIpApp.sntp.gmtOffset=0 setting on the polycom
phone provisioning template and I see that it has a variable for
${DSTOFFSET} in the template I tried adding dstoffset = -28800 to
users.conf and sip.conf under both general and individual users but can't
get the setting to set. Where/How do I set the variable?
Thanks,
Bryan
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Re: [asterisk-users] DST offset
Sorry I incorrectly typed my email. Let me correct, dstoffset should be
tzoffset.
Hello,
I am trying to set the tcpIpApp.sntp.gmtOffset=0 setting on the polycom
phone provisioning template and I see that it has a variable for
${TZOFFSET} in the template I tried adding tzoffset = -28800 to
users.conf and sip.conf under both general and individual users but can't
get the setting to set. Where/How do I set the variable?
Thanks,
Bryan
On Mon, Feb 25, 2013 at 10:44 AM, Bryan Anderson shadow...@gmail.comwrote:
Hello,
I am trying to set the tcpIpApp.sntp.gmtOffset=0 setting on the polycom
phone provisioning template and I see that it has a variable for
${DSTOFFSET} in the template I tried adding dstoffset = -28800 to
users.conf and sip.conf under both general and individual users but can't
get the setting to set. Where/How do I set the variable?
Thanks,
Bryan
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Re: [asterisk-users] DST offset
ok, so after digging through google and the wiki and not finding anything I
went through the code for res_phoneprov and found my answer. in users.conf
set timezone= to the full name of the time zone.
-Bryan Anderson
On Mon, Feb 25, 2013 at 1:35 PM, Bryan Anderson shadow...@gmail.com wrote:
Sorry I incorrectly typed my email. Let me correct, dstoffset should be
tzoffset.
Hello,
I am trying to set the tcpIpApp.sntp.gmtOffset=0 setting on the polycom
phone provisioning template and I see that it has a variable for
${TZOFFSET} in the template I tried adding tzoffset = -28800 to
users.conf and sip.conf under both general and individual users but can't
get the setting to set. Where/How do I set the variable?
Thanks,
Bryan
On Mon, Feb 25, 2013 at 10:44 AM, Bryan Anderson shadow...@gmail.comwrote:
Hello,
I am trying to set the tcpIpApp.sntp.gmtOffset=0 setting on the
polycom phone provisioning template and I see that it has a variable for
${DSTOFFSET} in the template I tried adding dstoffset = -28800 to
users.conf and sip.conf under both general and individual users but can't
get the setting to set. Where/How do I set the variable?
Thanks,
Bryan
--
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